Hello, I have an AAH-1.5 with a TMD400P with four lines, 8 Grandstream GXP-2000 phones, I am having echo issues on the GXP-2000 side. Here is what I have tried so far: The server has everything in the bios turned off except what is needed, USB, LPT, Serial etc,etc. I have uncommented Echo Suppresion in zconfig.h and shutdown and turned back on the asterisk box. I have updated the phones to 1.0.12 firmware, I have echotraining=800, echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using Mark2 as the echo suppresion and still I have echo. All the phones have been wired straight to the cisco 2950 switch and all cables have been tested and found to be good. I am completely at a loss at this point as to where to start looking and working to fix the problem. I would like to switch from Mark2 to MG1 but I don't know how I would acomplish that with AAH. I have played with the rx and tx gain but after reading multiple docs on it am still unsure how this would help and how to adjust it using /usr/bin/ztmonitor 1 -v. If anybody could point me in a new direction or something else to look at or something more to read that I may have missed I would be very appreciative. Thanks for any help, Jon
This pretty much helped me with the rxgains and txgains: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html you don't use ztmonitor directly to modify the rx and txgains, you just use it as a meter to make modifications to the zapata.conf file. Moj Jon Reynolds wrote:> Hello, > > I have an AAH-1.5 with a TMD400P with four lines, 8 Grandstream GXP-2000 > phones, I am having echo issues on the GXP-2000 side. > > Here is what I have tried so far: > > The server has everything in the bios turned off except what is needed, > USB, LPT, Serial etc,etc. > > I have uncommented Echo Suppresion in zconfig.h and shutdown and turned > back on the asterisk box. > > I have updated the phones to 1.0.12 firmware, I have echotraining=800, > echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using > Mark2 as the echo suppresion and still I have echo. > > All the phones have been wired straight to the cisco 2950 switch and all > cables have been tested and found to be good. > > I am completely at a loss at this point as to where to start looking and > working to fix the problem. I would like to switch from Mark2 to MG1 but > I don't know how I would acomplish that with AAH. I have played with the > rx and tx gain but after reading multiple docs on it am still unsure how > this would help and how to adjust it using /usr/bin/ztmonitor 1 -v. > > If anybody could point me in a new direction or something else to look > at or something more to read that I may have missed I would be very > appreciative. > > Thanks for any help, > > Jon > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Mojo <mojo@horanappraisals.com> Office Manger, Horan & Company, LLC (907) 747-6666 x112
Jon Reynolds wrote:> Hello, > > I have an AAH-1.5 with a TMD400P with four lines, 8 Grandstream GXP-2000 > phones, I am having echo issues on the GXP-2000 side. > > Here is what I have tried so far: > > The server has everything in the bios turned off except what is needed, > USB, LPT, Serial etc,etc. > > I have uncommented Echo Suppresion in zconfig.h and shutdown and turned > back on the asterisk box. > > I have updated the phones to 1.0.12 firmware, I have echotraining=800, > echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using > Mark2 as the echo suppresion and still I have echo.Is this correct? I do not believe having these echo parameters in sip.conf will achieve anything. They should be at the top of zapata.conf. Regards, Richard
On Wednesday, November 09, 2005 5:57 PM, Jon Reynolds wrote>Hello, > >I have an AAH-1.5 with a TMD400P with four lines, 8 >Grandstream GXP-2000 >phones, I am having echo issues on the GXP-2000 side.I have evaluated a similar setup as yours involving the Granstream 2000. I was able to isolate two sources of echo. 1. The Grandstream 2000 when the volume is up will cause echo because the microphone picks up the speaker on the handset. Don't even attempt to use speakerphone as you will cause full echo that will drive the remote party nuts. This problem is specific to the phone and doesn't relate to Asterisk. (Perhaps a newer firmware will resolve this?)> >Here is what I have tried so far: > >The server has everything in the bios turned off except what >is needed, >USB, LPT, Serial etc,etc. > >I have uncommented Echo Suppresion in zconfig.h and shutdown >and turned >back on the asterisk box. > >I have updated the phones to 1.0.12 firmware, I have echotraining=800, >echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using >Mark2 as the echo suppresion and still I have echo.2. Try the following settings in your zapata.conf. These seem to work well for me. echocancel=yes echocancelwhenbridged=yes echotraining=yes ; Use ztmonitor to adjust your gain to levels that work for you. rxgain=-4.0 txgain=-4.0> >All the phones have been wired straight to the cisco 2950 >switch and all >cables have been tested and found to be good. > >I am completely at a loss at this point as to where to start >looking and >working to fix the problem. I would like to switch from Mark2 >to MG1 but >I don't know how I would acomplish that with AAH. I have >played with the >rx and tx gain but after reading multiple docs on it am still >unsure how >this would help and how to adjust it using /usr/bin/ztmonitor 1 -v.When you place a call outbound, launch it and watch your gain as you speak. If you can humm a tone at around normal speaking voice to the far side, you can adjust the tx gain up or down to get it about halfway. Have the far end party do the same for the rx gain. It is trial and error. I was surprised to find that my setup worked best by turning the gain down. Check out this link for more info: http://www.voip-info.org/wiki/view/Asterisk+x100p+echotraining> >If anybody could point me in a new direction or something else to look >at or something more to read that I may have missed I would be very >appreciative. > >Thanks for any help, > >JonBTW, Digium recently released a new card with hardware-based echo cancellation. It may be worth a try. http://www.digium.com/index.php?menu=product_detail&category=hardware&pr oduct=TE411P&tab=details You may still hear echo at the first moment a call is placed, but it should completely disappear in a few seconds. -- Shawn